[Asterisk-Users] VoIP route SIP calls to provider

[iMRAN]

Dear Pros,

Can anyone be kind enough to guide me to route calls to my SIP carrier.

I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.

SIP.conf

[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729

[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833

[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833

extension.conf

[general]
static=yes
writeprotect=yes

[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000

[international]
ignorepat => 88
exten=> _1N1NXXNXXXXXX,1,Dial ???????

[internal]
include => local-sip

[local-sip]
exten => 1000,1,Dial(${PHONE1},40,t)
exten => 1000,2,Hangup

exten => 2000,1,Dial(${PHONE2},40,t)
exten => 2000,2,Hangup

exten => 1001,1,Dial(${PHONE3},40,t)
exten => 1001,2,Hangup

i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.

the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..

my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?

last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.

I thankyou all for reading this mail and i hope someone will be kind
enough to help.

Best regards,

Imran